It is often desirable to measure the time delay of a signal through an audio system. For example, it may be necessary to measure the difference in time delay as an audio signal passes through each of a plurality of speakers compared to a reference audio signal when trying to ensure synchronization between the speakers. Traditionally, time delay measurement techniques in an audio system involve using a known impulse signal. One method involves performing a cross-correlation between a transmitted impulse signal and the recorded audio signal. This method involves a training period while the adaptive algorithm adapts to the audio characteristics of the room, and requires calibration tones or known reference tones. Other methods include use of time-domain reflectometry where a pulse or a short sine wave burst is transmitted from the audio system. Measurements are then made of the timing of the return echo. These methods are susceptible to ambient noise and multi-modal reverberation and/or echo in a room. As a result, the recorded audio signal or return echo signal is not an exact replica of the original transmitted signal.
Also, adaptive filters are used for echo cancellation. In certain applications, such as in the case of TV set top boxes, the echo is delayed by an amount of time that exceeds the capacity of the adaptive filter. Increasing the filter size is not practical for digital signal processing reasons.